similar to: Voice mail and caller ID

Try this for CHAN_SIP: same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a mailbox defined log into it If you are using PJSIP it's more complex same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer same =>

On Fri, 12 Jun 2015 11:49:05 -0700 John Kiniston <johnkiniston at gmail.com> wrote: > Try this for CHAN_SIP: > > same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer > same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the > mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we > have a mailbox defined log into it Perfect.

On 12/2/19 11:52 AM, Joshua C. Colp wrote: > So I know that AMI is listening and I can talk to it.  Here is the > main log" > > [Nov 27 06:16:00] VERBOSE[101155] asterisk.c: Remote UNIX connection > [Nov 27 06:16:00] VERBOSE[101245] asterisk.c: Remote UNIX connection > disconnected > [Nov 27 06:16:01] VERBOSE[101244] manager.c: Manager

I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is working fine with a few minor tweaks except outgoinf fax. Incoming works fine. I do outgoing faxing through an AMI call. Here is the output from the security log: [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

On 12/3/19 3:04 PM, Joshua C. Colp wrote: > >     The AMI command, after the login, looks like this: > > > >     Action: Originate > >     Channel: SIP/outgoing/%%(destination)s > >     Context: LocalSets > >     CallerID: Vybe Consulting Inc Fax Service <5555551212> > >     Exten: sendfax > >   

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%)

I am having troubles with sending faxes. I hope someone can help me work out a better method. Basically we have a special address that our users can send to. It winds up on our Asterisk server which runs a Python script that parses the message for attachments and the phone number from the recipient address. The attachments are converted to TIFF and stored in a folder with various information

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

Not specific to CentOS but I know you guys would be really helpful anyhow. Basically, I have a file which has been editted in the past very similarly to the hosts file only now I want to use it as a hosts file and need to run some fancy sed to massage the data into shape. Currently, the data in the file is in the form of <ip address> <tab> <short hostname> <space>

Hi Folks, Like to get some advice on module organization. I have a system that has a few different components. One or more components can be installed on a host. The hosts also have a "type" dimension. This "type" dimension could (should) ideally be used to configure the system (fill in a bunch of variables). Problem is, I''m not sure how the best way to do

I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.

On Tue, 30 Aug 2016 10:39:14 -0400 Eric Wieling <ewieling at nyigc.com> wrote: > The dialplan below cannot go to voicemail, either something else is Of course not. It's the individual extensions that have voice mail. I have a similar problem when one of those destinations is a cell phone but I know that there is no Asterisk solution for that problem. If the cell phone answers and

On Thu, 4 Aug 2016 14:03:39 +0100 Nabeel <nabeelshikder at gmail.com> wrote: > I should add, a password is *always* asked if a password has been set. > There isn't a way to bypass that. Then something is wrong. http://darcy.vex.net/star98.mp3 -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net

I know that it runs on other systems but do other ports get the same attention? I have been running it on a NetBSD server for about a year now and while it mostly works it just crashes from time to time with no explanation or core dump. I have improved the situation by expanding my intrusion detection but it still stops every few days or so. I have a cron job that tests for it and restarts it

On 4 August 2016 at 13:18, D'Arcy J.M. Cain <darcy at vex.net> wrote: > > Let's get this straight. You call yourself from any phone in the world > and press '*' while listening to the message, you wind up in your own > mailbox and you believe that means that you don't need a password? Do > you think that the phone system somehow knows that it is you

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

I am hoping someone else has seen this and can offer a solution or at least a direction to investigate. I am running 11.23. Most of my clients are fine but one has a strange behaviour. He has a Grandstream HT701 like most of my clients who use an ATA. He can make call and they are crystal clear. However, when he tries to use phone menus ("dial 234 for John Doe" for example) it